• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.3B
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• pp.144-151
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• 2008

In this paper, we propose a video streaming server that guarantees a certain level of quality when a server should serve video streaming service to multiple heterogenous clients simultaneously with TCP transport. If each heterogeneous client requests video streaming service in according to its own requirement such as bitrate of content and these requests are accepted by a server, then TCP flows for each video streaming session fairly share limited uplink bandwidth of the server. At this time, because TCP's bandwidth fair-share characteristics can result in bandwidth shrinkage of higher bitrate video streaming session, the client of higher bitrate video may suffer sluggish playback which is related with streaming QoS degradation. To tackle this problem, our proposed server system uses multiple TCP connections adaptively for each video streaming session depending on the anticipated status of the client playout buffer. Simulation results show that our proposed algorithm can successfully reduce the occurrence of playout buffer underrun and enhance streaming quality for whole video clients.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.11B
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• pp.784-791
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• 2005

To provide VoD Service on the Internet VoD server is required to have a large amount of system resource and network bandwidth. Therefore, overlay Multicast schemes are considered as suitable alternatives but they also have some drawbacks to support on-demand media services. In this paper, we propose an overlay multicast based on-demand media service scheme which could exploit server's system resource and its network bandwidth efficiently. Proposed scheme uses shared buffer of clients involving the relay of traffic and Patching while it gives no restrictions to this scheme compared with unicast. Our simulation results show that proposed scheme can support more user than unicast and improve the network performance at the same time.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2006.05a
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• pp.729-732
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• 2006

This paper proposes and implements consistency algorithm that is serviced streaming data efficiently in VOD system. The media data is loaded into each node by Round Robin method. The barrier mechanism is changed into the minimum data factor(SH. GOP) of media data in this paper. In addition in order to fast media data service, the additional features are implemented in the consistency algorithm. Additional feature of the consistency algorithm is added. First, time synchronization algorithm is added the suggested consistency algorithm. Second, the prepaging mechanism supports efficient buffering service. I experimented the suggested consistency algorithm on two DSM systems.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.12 no.3
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• pp.509-516
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• 2008

Recently, the operator modules to control external devices are concerned about automatic management system to process continuously changed signals. These signals are the stream data of which characteristics are several numbers. a short report interval and asynchronous report time. It is necessary that the system brings about high accuracy and real time process for stream data. The typical queries of these systems consist of the current query to search the latest signal value, the snapshot query at a past time, the sliding window query from a past time to current. In this paper, we propose the efficient method to manage the above signals by using a file structured database in small-size operating systems. We also propose a query model to accommodate various queries including the sliding window query. The file database in the QNX adopts a delta version and a shared memory buffering method for the resource limit of a small storage and a low computing power.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.18 no.6
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• pp.779-792
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• 1993

In a switching node or an ATM multiplexer of the ATM network, a good bandwidth utilization can be achieved by the priority control using the 1-bit(Cell Loss priority) in ATM cell header. In this paper, the mixed mechanism is proposed to make up for shortcomings of existing space priority control mechanisms and to decrease the loss probability of high priority cell and its performance is analyzed about the cell loss probability. To estimate the performance of proposed mixed mechanism, its cell loss probability is compared with those of non-priority mechanism, push-out mechanism and partial buffer sharing mechanism. The cell loss probability is analyzed using a M/D/1/N modeling and a 2-state MMPP/D/1/N modeling and also comparison between two modelings is made. To verify this result of numerical analysis, the computer simulation is performed for each mechanism using the simulation language, SIMSRIPT II.5.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.9 no.1
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• pp.45-57
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• 2005

In this paper, we extend the performance of bidirectional TCP connection over end-to-end network that uses transfer rate-based flow and congestion control. The sharing of a common buffer by TCP packets and acknowledgement has been known to result in an effect called ack compression, where acks of a connection arrive at the source bunched together, resulting in unfairness and degraded throughput. The degradation in throughput due to bidirectional traffic can be significant. Even in the simple case of symmetrical connections with adequate window size, the connection efficiency is improved about 20% for three levels of background traffic 2.5Mbps, 5.0Mbps and 7.5Mbps. Otherwise, the throughput of jitter is reduced about 50% because round trip delay time is smaller between source node and destination node. Also, we show that throughput curve is improved with connection rate algorithm which is proposed for TCP congetion avoidance as a function of aggressiveness threshold for three levels of background traffic 2.5Mbps, 5Mbps and 7.5Mbps. By analyzing the periodic bursty behavior of the source IP queue, we derive estimated for the maximum queue size and arrive at a simple predictor for the degraded throughput, applicable for relatively general situations.

• The Journal of the Korea institute of electronic communication sciences
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• v.1 no.2
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• pp.102-109
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• 2006

In this paper, we improve the performance of bidirectional TCP connection over end-to-end network that uses transfer rate-based flow and congestion control. The sharing of a common buffer by TCP packets and acknowledgement has been known to result in an effect called ack compression, where acks of a connection arrive at the source bunched together, resulting in unfairness and degraded throughput. The degradation in throughput due to bidirectional traffic can be significant. For example, even in the simple case of symmetrical connections with adequate window size, the connection efficiency is improved about 20% for three levels of background traffic 2.5Mbps, 5.0Mbps and 7.5Mbps. Otherwise, the throughput of jitter is reduced about 50% because round trip delay time is smaller between source node and destination node. Also, we show that throughput curve is improved with connection rate algorithm which is proposed for TCP congestion avoidance as a function of aggressiveness threshold for three levels of background traffic 2.5Mbps, 5Mbps and 7.5Mbps.

• Journal of Digital Contents Society
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• v.3 no.1
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• pp.47-57
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• 2002

Recently, wireless networks have been pursuing multimedia data service as voice, data, image, video and various form of data according to development of information communication technology. It guarantees cell delivery delay of real time data in efficient real time multimedia data transfer. Also, it minimizes cell loss rate of non-real time multimedia data. In the wireless ATM, there are based on Asynchronous Transfer Mode(ATM). It implies that there are various service with difficult transmission rates and qualities in the wireless communication network. As a result, it is important to find out the ways to guarantee the Quality of Service(QoS) for each kind of traffic in wireless network. In this thesis, we propose an improved TCRM scheduling algorithms for transmission real-time multimedia data service in wireless ATM Networks. We appear real time multimedia scheduling policy that apply each different method to uplink and downlik to wireless ATM network. It can guarantee QoS requirements for each real time data and non-real time data. It also deals the fairness problem for sharing the scarce wireless resources. We solve fault of TCRM as inefficient problem of non-real data by using arbitrary transmission speed and RB(Reservation Buffer) through VC(Virtual Control) and BS(Base Station).

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.11C
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• pp.1112-1123
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• 2002

Providing a multimedia service requires efficient network bandwidth allocation scheduling. Typically, transmission of information with bursty characteristics, such as a video data stream, utilizes some type of smoothing technique that reduces the transmission rate fluctuation. While capable of a drastic reduction in the peak transmission rate, even the optimum smoothing algorithm proposed to date, however, is rather inadequate for multi-stream applications that share network bandwidth since such an algorithm has been designed and applied toward handling each independent stream. In this paper, we proposed an efficient bandwidth allocation algorithm that addresses these shortcomings. The proposed bandwidth allocation algorithm is divided into two steps, a static bandwidth allocation and a dynamic bandwidth allocation. In the former case, the peak rate reduction is achieved by maintaining the accumulated data in the user buffer at a high level, whereas the concept of real demand factor is employed to meet the needs of multi-stream bandwidth allocation in the latter case, Finally, the proposed algorithm's performance was verified with ExSpect 6.41, a Petri net simulation tool.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.1A
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• pp.44-53
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• 2010

This paper proposes a traffic management policy for delay control in the wireless networks. The so-called EDD(Earliest Due Date) scheme is adopted as the packet scheduling policy, so that the service provision is performed in the order of the transmission urgency of the backlogged packets. In addition, we derive a formula to determine the contention window, one of the MAC parameters, with the goal of minimizing the non-work conserving characteristics of the traditional MAC scheme. This method eliminates the burden of the class-wise parameter settings which is typically required for the priority control. Simulations are performed to show the validity of the proposed scheme in comparison with the policy that adopts the class-level queue management such as the IEEE 802.11e standard. Smaller delays and higher rates of delay guarantees are observed throughout the experiments.